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kodi: add PR5682
Signed-off-by: Stephan Raue <stephan@openelec.tv>
This commit is contained in:
Stephan Raue
parent
fbed78c45f
commit
b444bcbeff
141
packages/mediacenter/kodi/patches/kodi-999.92-PR5682.patch
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141
packages/mediacenter/kodi/patches/kodi-999.92-PR5682.patch
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@ -0,0 +1,141 @@
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From 970ff0341f05a0adfe3ca79e5fba26a7070a12b7 Mon Sep 17 00:00:00 2001
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From: fritsch <Peter.Fruehberger@gmail.com>
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Date: Sun, 9 Nov 2014 17:46:10 +0100
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Subject: [PATCH 2/2] AESinkALSA: Allow fragmentation, e.g. even multiply of
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PeriodSize
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---
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xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp | 66 ++++++++++++++++++++++-------
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xbmc/cores/AudioEngine/Sinks/AESinkALSA.h | 4 +-
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2 files changed, 53 insertions(+), 17 deletions(-)
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diff -Naur kodi-14-2d88a9a/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp kodi-14-2d88a9a.patch/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp
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--- kodi-14-2d88a9a/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp 2014-11-09 22:28:18.035651639 +0100
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+++ kodi-14-2d88a9a.patch/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp 2014-11-10 00:35:57.351393193 +0100
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@@ -39,6 +39,8 @@
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#include "utils/AMLUtils.h"
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#endif
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+
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+#define AE_MIN_PERIODSIZE 256
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#define ALSA_CHMAP_KERNEL_BLACKLIST
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#define ALSA_OPTIONS (SND_PCM_NO_AUTO_FORMAT | SND_PCM_NO_AUTO_CHANNELS | SND_PCM_NO_AUTO_RESAMPLE)
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@@ -86,7 +88,9 @@
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m_formatSampleRateMul(0.0),
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m_passthrough(false),
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m_pcm(NULL),
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- m_timeout(0)
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+ m_timeout(0),
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+ m_fragmented(false),
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+ m_originalPeriodSize(AE_MIN_PERIODSIZE)
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{
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/* ensure that ALSA has been initialized */
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if (!snd_config)
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@@ -479,7 +483,8 @@
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params += ",AES1=0x82,AES2=0x00";
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- if (format.m_sampleRate == 192000) params += ",AES3=0x0e";
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+ if (m_passthrough && format.m_channelLayout.Count() == 8) params += ",AES3=0x09";
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+ else if (format.m_sampleRate == 192000) params += ",AES3=0x0e";
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else if (format.m_sampleRate == 176400) params += ",AES3=0x0c";
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else if (format.m_sampleRate == 96000) params += ",AES3=0x0a";
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else if (format.m_sampleRate == 88200) params += ",AES3=0x08";
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@@ -806,7 +811,19 @@
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/* set the format parameters */
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outconfig.sampleRate = sampleRate;
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- outconfig.periodSize = periodSize;
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+
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+ /* if periodSize is too small Audio Engine might starve */
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+ m_fragmented = false;
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+ unsigned int fragments = 1;
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+ if (periodSize < AE_MIN_PERIODSIZE)
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+ {
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+ fragments = std::ceil((double) AE_MIN_PERIODSIZE / periodSize);
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+ CLog::Log(LOGDEBUG, "Audio Driver reports too low periodSize %d - will use %d fragments", (int) periodSize, (int) fragments);
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+ m_fragmented = true;
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+ }
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+
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+ m_originalPeriodSize = periodSize;
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+ outconfig.periodSize = fragments * periodSize;
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outconfig.frameSize = snd_pcm_frames_to_bytes(m_pcm, 1);
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m_bufferSize = (unsigned int)bufferSize;
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@@ -893,27 +910,45 @@
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}
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void *buffer = data[0]+offset*m_format.m_frameSize;
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- int ret = snd_pcm_writei(m_pcm, buffer, frames);
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- if (ret < 0)
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- {
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- CLog::Log(LOGERROR, "CAESinkALSA - snd_pcm_writei(%d) %s - trying to recover", ret, snd_strerror(ret));
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- ret = snd_pcm_recover(m_pcm, ret, 1);
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- if(ret < 0)
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- {
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- HandleError("snd_pcm_writei(1)", ret);
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- ret = snd_pcm_writei(m_pcm, buffer, frames);
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- if (ret < 0)
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+ unsigned int amount = 0;
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+ int64_t data_left = (int64_t) frames;
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+ int frames_written = 0;
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+
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+ while (data_left > 0)
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+ {
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+ if (m_fragmented)
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+ amount = std::min((unsigned int) data_left, m_originalPeriodSize);
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+ else // take care as we can come here a second time if the sink does not eat all data
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+ amount = (unsigned int) data_left;
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+
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+ int ret = snd_pcm_writei(m_pcm, buffer, amount);
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+ if (ret < 0)
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+ {
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+ CLog::Log(LOGERROR, "CAESinkALSA - snd_pcm_writei(%d) %s - trying to recover", ret, snd_strerror(ret));
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+ ret = snd_pcm_recover(m_pcm, ret, 1);
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+ if(ret < 0)
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{
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- HandleError("snd_pcm_writei(2)", ret);
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- ret = 0;
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+ HandleError("snd_pcm_writei(1)", ret);
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+ ret = snd_pcm_writei(m_pcm, buffer, amount);
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+ if (ret < 0)
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+ {
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+ HandleError("snd_pcm_writei(2)", ret);
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+ ret = 0;
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+ }
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}
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}
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- }
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- if ( ret > 0 && snd_pcm_state(m_pcm) == SND_PCM_STATE_PREPARED)
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- snd_pcm_start(m_pcm);
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+ if ( ret > 0 && snd_pcm_state(m_pcm) == SND_PCM_STATE_PREPARED)
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+ snd_pcm_start(m_pcm);
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- return ret;
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+ if (ret <= 0)
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+ break;
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+
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+ frames_written += ret;
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+ data_left -= ret;
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+ buffer = data[0]+offset*m_format.m_frameSize + frames_written*m_format.m_frameSize;
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+ }
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+ return frames_written;
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}
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void CAESinkALSA::HandleError(const char* name, int err)
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diff -Naur kodi-14-2d88a9a/xbmc/cores/AudioEngine/Sinks/AESinkALSA.h kodi-14-2d88a9a.patch/xbmc/cores/AudioEngine/Sinks/AESinkALSA.h
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--- kodi-14-2d88a9a/xbmc/cores/AudioEngine/Sinks/AESinkALSA.h 2014-11-09 22:28:18.036651638 +0100
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+++ kodi-14-2d88a9a.patch/xbmc/cores/AudioEngine/Sinks/AESinkALSA.h 2014-11-10 00:36:24.936426057 +0100
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@@ -90,6 +90,10 @@
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#endif
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static CALSAHControlMonitor m_controlMonitor;
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+ // support fragmentation, e.g. looping in the sink to get a certain amount of data onto the device
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+ bool m_fragmented;
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+ unsigned int m_originalPeriodSize;
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+
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struct ALSAConfig
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{
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unsigned int sampleRate;
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@ -1,29 +0,0 @@
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From 22477eaf1286dd28fb87552d35e9c8a5dbc09658 Mon Sep 17 00:00:00 2001
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From: Rudi <r.ihle@s-t.de>
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Date: Sat, 9 Aug 2014 18:09:03 +0200
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Subject: [PATCH] AESinkALSA: Set correct value for AES3 when using HBR mode
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According to the HDMI 1.4a document, the only acceptable value
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for AES3 in HBR mode is 0x09, which corresponds to a frame rate
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of 768000. Some sinks or drivers ignore this, but some do not.
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---
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xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp | 3 ++-
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1 file changed, 2 insertions(+), 1 deletion(-)
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diff --git a/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp b/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp
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index c608231..322b8de 100644
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--- a/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp
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+++ b/xbmc/cores/AudioEngine/Sinks/AESinkALSA.cpp
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@@ -156,7 +156,8 @@ void CAESinkALSA::GetAESParams(AEAudioFormat format, std::string& params)
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params += ",AES1=0x82,AES2=0x00";
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- if (format.m_sampleRate == 192000) params += ",AES3=0x0e";
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+ if (m_passthrough && format.m_channelLayout.Count() == 8) params += ",AES3=0x09";
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+ else if (format.m_sampleRate == 192000) params += ",AES3=0x0e";
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else if (format.m_sampleRate == 176400) params += ",AES3=0x0c";
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else if (format.m_sampleRate == 96000) params += ",AES3=0x0a";
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else if (format.m_sampleRate == 88200) params += ",AES3=0x08";
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--
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2.0.3
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@ -1,11 +0,0 @@
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--- xbmc-master-14-dae6f76/./xbmc/cores/AudioEngine/AESinkFactory.cpp.orig 2014-07-07 18:30:37.000000000 +0200
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+++ xbmc-master-14-dae6f76/./xbmc/cores/AudioEngine/AESinkFactory.cpp 2014-07-15 10:30:16.513471771 +0200
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@@ -134,7 +134,7 @@
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CLog::Log(LOGERROR, "Sink %s:%s returned invalid sample rate", driver.c_str(), device.c_str());
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else if (format.m_channelLayout.Count() == 0)
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CLog::Log(LOGERROR, "Sink %s:%s returned invalid channel layout", driver.c_str(), device.c_str());
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- else if (format.m_frames < 256)
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+ else if (format.m_frames < 96)
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CLog::Log(LOGERROR, "Sink %s:%s returned invalid buffer size: %d", driver.c_str(), device.c_str(), format.m_frames);
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else
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return sink;
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